Beyond QoS: Voice Quality Management for VoIP Networks

Why Standard QoS Isn't Sufficient for Voice
Service providers typically use three techniques to control and enhance QoS in packet-based networks: class of service (CoS), differentiated services (DiffServ), and Multi-Protocol Label Switching (MPLS). All three have one thing in common: they all attempt to minimize packet loss and jitter of the voice transport. While this is one piece of the puzzle, controlling packet loss and jitter does not prevent other impairments from degrading the voice.

CoS (IEEE 802.1p) is the usual means for providing QoS in Ethernet networks. With CoS, all packets are assigned a priority, and VoIP traffic generally receives the highest priority; thus, using CoS prioritization can minimize VoIP traffic packet loss.

DiffServ is a class-based IP QoS technique specified by the Internet Engineering Task Force (IETF). The DiffServ Control Point contained in the IP header is used to control the Per-Hop Behavior (PHB) of routers along the traffic's path from end to end. With VoIP traffic, the PHB is normally set for "Expedited Forwarding" to minimize latency and packet loss.

MPLS adds a separate 32-bit header to each packet which is used to create virtual label switched paths (LSPs). With LSPs, providers can segment, prioritize, and expedite traffic. Originally created as a means to implement virtual private networks (VPNs), the MPLS header also has a 3-bit QoS field that can be used to minimize packet loss and latency with VoIP traffic.

Measuring IP Voice Quality
While IP QoS techniques are useful in any network, VoIP requires more detailed voice quality management. The first challenge is to quantify voice quality so it can be monitored and managed.

In the public switched telephone network (PSTN), the mean opinion score (MOS) remains the benchmark for determining voice quality. Although the subjective nature of MOS testing is practical in the stable and predictable PSTN, the dynamic and "bursty" nature of IP traffic requires a different approach. For this reason, newer, objective techniques that can be automated are now available to calculate voice quality ratings based on measured values. Of these techniques, the E-Model specified in ITU-T Recommendation G.107 is the least intrusive and the most cost-effective. By measuring various parameters of the network and the speech, the E-Model estimates how the average user would likely rate a call on an equivalent MOS scale (see Table 1).

The R-Factor is calculated based on impairments such as a call's signal-to-noise ratio (SNR), level mismatch and codec distortion, talker and listener echo, and codec tolerance for packet loss. A detailed analysis of the R-Factor can be used to help isolate the root cause of persistent problems.

Customers have two sets of expectations regarding voice quality. In the PSTN, the expectation is for "toll quality" voice with a MOS rating of 4.0 or more. In the cellular phone network, the expectation is considerably lower (around MOS 3.2"3.5). Of course, cellular customers willingly tolerate this "fair" voice quality for the convenience of mobility.

There are three reasons why it's so difficult to achieve a MOS of 4.0 or more in an IP network. First is the use of low bit-rate encoding in many VoIP networks as a way of minimizing bandwidth demand. For commonly-used, low bit-rate codecs such as G.729a (8 kbps) the highest possible MOS score is only 3.7, even under the best of network conditions. Additionally, packet loss dramatically affects the speech quality of the low bit rate codecs. In contrast, the G.711 codec (64 kbps) employed in the PSTN can tolerate packet loss very well and has a starting MOS score above 4.0.

The second reason is that IP voice quality inevitably deteriorates as the number of subscribers grows and/or the traffic load increases. Adding capacity mitigates the problem, of course, but the bursty nature of IP traffic makes it inevitable that the network will experience periods of congestion so that despite QoS implementation, users experience diminished voice quality from packet loss, delay and jitter.

The third reason is that IP voice quality is frequently impaired by noise, distortion, echo, and mismatched volume levels. Many IP networks, by design, simply lack the provisions to eliminate these impairments.

Voice Quality Impairments
Now, let's look at IP voice quality impairments in more detail. Figure 1 depicts a typical end-to-end VoIP network. Each network segment (from the subscriber premises through the core and beyond) is assumed to offer some form of QoS.